Release packages

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Important

Merging this pull request will create these releases

livekit 0.7.40 (2026-05-14)

Fixes

• feat: add scalability mode for AV1/VP9. - feat: add scalability mode for AV1/VP9. #1076 (@cloudwebrtc)
• Add LIVEKIT_PREFERRED_HW_ENCODER to prefer nvenc or vaapi hardware video encoding when both are available.
• Relocate unrelated types out of livekit-protocol

Get WebRTC ADM into Rust - #1037 (@xianshijing-lk)

This PR introduces platform audio device management via WebRTC's Audio Device Module (ADM).

Features

• ADM Proxy: New AdmProxy class that switches between Dummy ADM (synthetic mode) and Platform ADM (real audio I/O)
• PlatformAudio API: High-level Rust API for microphone capture and speaker playout with AEC/AGC/NS
• Device enumeration: List and select recording/playout devices by index or GUID
• Mode switching: Seamlessly switch between synthetic mode (FFI callbacks) and platform mode (native speakers) while audio is active
• FFI platform audio support: Expose platform audio device enumeration and selection through livekit-ffi
• Audio processing: Configure echo cancellation, noise suppression, and auto gain control with platform-specific defaults (hardware on iOS, software elsewhere)

Audio Modes

Mode	Recording	Playout	Use Case
Synthetic	NativeAudioSource	Dummy ADM + FFI	Unity audio, agents
Platform	Platform ADM mic	Platform ADM speakers	VoIP with AEC

API

// Create PlatformAudio for microphone/speaker access
let audio = PlatformAudio::new()?;

// Enumerate and select devices
for i in 0..audio.recording_devices() as u16 {
    println!("Mic {}: {}", i, audio.recording_device_name(i));
}
audio.set_recording_device(0)?;

// Create audio track for publishing
let track = LocalAudioTrack::create_audio_track("mic", audio.rtc_source());

livekit-api 0.4.22 (2026-05-14)

Fixes

• Proper client SDK - Proper setting of client SDK #1081 (@stephen-derosa)

livekit-protocol 0.7.7 (2026-05-14)

Features

• Relocate unrelated types out of livekit-protocol

livekit-ffi 0.12.57 (2026-05-14)

Fixes

• feat: add scalability mode for AV1/VP9. - feat: add scalability mode for AV1/VP9. #1076 (@cloudwebrtc)
• Add LIVEKIT_PREFERRED_HW_ENCODER to prefer nvenc or vaapi hardware video encoding when both are available.
• Reword audio filter logs to be less confusing - Reword audio filter logs to be less confusing #1092 (@1egoman)

Get WebRTC ADM into Rust - #1037 (@xianshijing-lk)

This PR introduces platform audio device management via WebRTC's Audio Device Module (ADM).

Features

• ADM Proxy: New AdmProxy class that switches between Dummy ADM (synthetic mode) and Platform ADM (real audio I/O)
• PlatformAudio API: High-level Rust API for microphone capture and speaker playout with AEC/AGC/NS
• Device enumeration: List and select recording/playout devices by index or GUID
• Mode switching: Seamlessly switch between synthetic mode (FFI callbacks) and platform mode (native speakers) while audio is active
• FFI platform audio support: Expose platform audio device enumeration and selection through livekit-ffi
• Audio processing: Configure echo cancellation, noise suppression, and auto gain control with platform-specific defaults (hardware on iOS, software elsewhere)

Audio Modes

Mode	Recording	Playout	Use Case
Synthetic	NativeAudioSource	Dummy ADM + FFI	Unity audio, agents
Platform	Platform ADM mic	Platform ADM speakers	VoIP with AEC

API

// Create PlatformAudio for microphone/speaker access
let audio = PlatformAudio::new()?;

// Enumerate and select devices
for i in 0..audio.recording_devices() as u16 {
    println!("Mic {}: {}", i, audio.recording_device_name(i));
}
audio.set_recording_device(0)?;

// Create audio track for publishing
let track = LocalAudioTrack::create_audio_track("mic", audio.rtc_source());

libwebrtc 0.3.33 (2026-05-14)

Fixes

• feat: add scalability mode for AV1/VP9. - feat: add scalability mode for AV1/VP9. #1076 (@cloudwebrtc)
• Add LIVEKIT_PREFERRED_HW_ENCODER to prefer nvenc or vaapi hardware video encoding when both are available.
• Relocate unrelated types out of livekit-protocol

Get WebRTC ADM into Rust - #1037 (@xianshijing-lk)

This PR introduces platform audio device management via WebRTC's Audio Device Module (ADM).

Features

• ADM Proxy: New AdmProxy class that switches between Dummy ADM (synthetic mode) and Platform ADM (real audio I/O)
• PlatformAudio API: High-level Rust API for microphone capture and speaker playout with AEC/AGC/NS
• Device enumeration: List and select recording/playout devices by index or GUID
• Mode switching: Seamlessly switch between synthetic mode (FFI callbacks) and platform mode (native speakers) while audio is active
• FFI platform audio support: Expose platform audio device enumeration and selection through livekit-ffi
• Audio processing: Configure echo cancellation, noise suppression, and auto gain control with platform-specific defaults (hardware on iOS, software elsewhere)

Audio Modes

Mode	Recording	Playout	Use Case
Synthetic	NativeAudioSource	Dummy ADM + FFI	Unity audio, agents
Platform	Platform ADM mic	Platform ADM speakers	VoIP with AEC

API

// Create PlatformAudio for microphone/speaker access
let audio = PlatformAudio::new()?;

// Enumerate and select devices
for i in 0..audio.recording_devices() as u16 {
    println!("Mic {}: {}", i, audio.recording_device_name(i));
}
audio.set_recording_device(0)?;

// Create audio track for publishing
let track = LocalAudioTrack::create_audio_track("mic", audio.rtc_source());

webrtc-sys 0.3.31 (2026-05-14)

Fixes

• chore: bump libwebrtc version to webrtc-51ef663
• Add LIVEKIT_PREFERRED_HW_ENCODER to prefer nvenc or vaapi hardware video encoding when both are available.

fix: fix LICENSE.md generation in webrtc build scripts

• Add fix_license_json_parsing.patch to handle GN warnings in JSON output
• Enable add_licenses.patch for iOS and Android builds (was commented out)
• Restore LICENSE.md copy in iOS build script (regression from Improve WebRTC build scripts and add external_audio_source patch #1053)

The license generation script was failing because gn desc --format=json
outputs warnings before the JSON when certain build args trigger deprecation
notices. The new patch strips non-JSON content before parsing.

Get WebRTC ADM into Rust - #1037 (@xianshijing-lk)

This PR introduces platform audio device management via WebRTC's Audio Device Module (ADM).

Features

• ADM Proxy: New AdmProxy class that switches between Dummy ADM (synthetic mode) and Platform ADM (real audio I/O)
• PlatformAudio API: High-level Rust API for microphone capture and speaker playout with AEC/AGC/NS
• Device enumeration: List and select recording/playout devices by index or GUID
• Mode switching: Seamlessly switch between synthetic mode (FFI callbacks) and platform mode (native speakers) while audio is active
• FFI platform audio support: Expose platform audio device enumeration and selection through livekit-ffi
• Audio processing: Configure echo cancellation, noise suppression, and auto gain control with platform-specific defaults (hardware on iOS, software elsewhere)

Audio Modes

Mode	Recording	Playout	Use Case
Synthetic	NativeAudioSource	Dummy ADM + FFI	Unity audio, agents
Platform	Platform ADM mic	Platform ADM speakers	VoIP with AEC

API

// Create PlatformAudio for microphone/speaker access
let audio = PlatformAudio::new()?;

// Enumerate and select devices
for i in 0..audio.recording_devices() as u16 {
    println!("Mic {}: {}", i, audio.recording_device_name(i));
}
audio.set_recording_device(0)?;

// Create audio track for publishing
let track = LocalAudioTrack::create_audio_track("mic", audio.rtc_source());

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